On 04/28/2011 10:42 AM, wisni...@onet.eu wrote:
> Hi
> How UAS should react having received OPTIONS, out of the dialog, with "To"
> field containing tag? Is "8.2.6.2 Headers and Tags" from RFC3261 proper
> requirement for this isue?
> In my case UAS respond with 481Unknown Dialog.
> Regards
> Bar
On 03/31/2011 04:13 PM, Nitin Kapoor wrote:
> Dear All,
>
> I am facing the issue with one of my client, where my Termination is sending
> 183 with SDP but my UAC is unable to hear any destination country ringback.
>
>
Hello,
Although this is very weird, but does the SBC also send a 180 Ringin
On 10/13/2010 12:26 PM, SIP Satan wrote:
> Load balancing seems to be the valid purpose for this behaviour than
> billing(statitics) related purposes.
>
>
Well , to be frank , load balancing is a little difficult because you
already have route headers from the other hops, so requests are
loos
On 10/13/2010 10:40 AM, sathwikh gn wrote:
> Hello ,
> I am very new to the SIP. I just read in RFC 3261 that stateless proxy can
> add Record Route. Since the Stateless proxy does not maintain any state ,
> what is the significance of Record Route added by Stateless Proxy.
>
> Thanks
>
Hello
Vinod Parameswaran wrote:
> Hi Marius,
>
> Thanks. I am not sure I quite understood what you mentioned in your previous
> mail.
> Do you mean use a customized stack at the transport layer in order to make
> multiple connections?
> In that case, I am afraid I do not understand why I need to tweak
Vinod Parameswaran wrote:
> Hi,
>
> does anyone know of a SIP opensource stack that has been ported to Symbian
> and supports multi-threading for applications? I am aware of PJSIP which has
> been ported to Symbian, but does not support multi-threading for applications.
>
> Any suggestions would
g
changes/annotations to a draft ?
Marius
From: Paul E. Jones [pau...@packetizer.com]
Sent: Monday, August 16, 2010 6:45 PM
To: Marius Zbihlei
Cc: sip-...@ietf.org; sip-implementors@lists.cs.columbia.edu; 'Gonzalo
Salgueiro'; 'Hadriel Kaplan';
Paul Kyzivat wrote:
> $...@r\/|>r!`/@ wrote:
>
>> Hi All,
>>
>> Can you please refer me a write up which explains how a proxy handles
>> transport switching scenarios if required which frwding request.
>>
>
> Can you be more specific about your question?
>
> Each hop is a separate sip trans
t
8.3. Session Expiration
When the current time equals or passes the session expiration for a
session, the proxy MAY remove associated call state, and MAY free any
resources associated with the call. Unlike the UA, it MUST NOT send
a BYE.
>> -Original Message-
>> From: marius zbihle
Paul E. Jones wrote:
> Folks,
>
>
>
> Gonzalo and I produced an Internet Draft aiming at trying to bring some
> consistency to the way in which SIP user agents implement an OPTIONS "ping"
> procedure. It seems that a very large number of vendors do this, but
> unfortunately, there seems to be li
Aaron Clauson wrote:
> What sort of SIP message processing throughput should be expected from a SIP
> stack operating in a proxy role, no media processing, and on a commodity x86
> server (for arguments sake a 3GHz quad core Xeon or anything in that
> ballpark)? 1k, 10k, 100k, >100k messages per s
Alex Balashov wrote:
> On 07/30/2010 04:59 AM, WORLEY, Dale R (Dale) wrote:
>
>
>> While 3261 systems should interoperate correctly with 2543 systems,
>> 2543 systems are considered obsolete these days.
>>
>
> I think the issue is that there is a proxy inserting an RR header with
> the 'lr
Alex Bakker wrote:
> Hello,
>
> When I make a call from a Siemens handset to SJPhone (softphone), I notice
> something strange in the "To:" field in the INVITE request.
>
> This field looks like this:
>
> To:
>
> Is this really a valid notation?? I am writing a parser that reads these
>
Hello A
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